@@ -67,6 +67,7 @@ struct q6asm_dai_rtd {
uint16_t bits_per_sample;
uint16_t source; /* Encoding source bit mask */
struct audio_client *audio_client;
+ uint32_t stream_id;
uint16_t session_id;
enum stream_state state;
};
@@ -184,7 +185,7 @@ static void event_handler(uint32_t opcode, uint32_t token,
switch (opcode) {
case ASM_CLIENT_EVENT_CMD_RUN_DONE:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- q6asm_write_async(prtd->audio_client,
+ q6asm_write_async(prtd->audio_client, prtd->stream_id,
prtd->pcm_count, 0, 0, NO_TIMESTAMP);
break;
case ASM_CLIENT_EVENT_CMD_EOS_DONE:
@@ -194,7 +195,7 @@ static void event_handler(uint32_t opcode, uint32_t token,
prtd->pcm_irq_pos += prtd->pcm_count;
snd_pcm_period_elapsed(substream);
if (prtd->state == Q6ASM_STREAM_RUNNING)
- q6asm_write_async(prtd->audio_client,
+ q6asm_write_async(prtd->audio_client, prtd->stream_id,
prtd->pcm_count, 0, 0, NO_TIMESTAMP);
break;
@@ -203,7 +204,7 @@ static void event_handler(uint32_t opcode, uint32_t token,
prtd->pcm_irq_pos += prtd->pcm_count;
snd_pcm_period_elapsed(substream);
if (prtd->state == Q6ASM_STREAM_RUNNING)
- q6asm_read(prtd->audio_client);
+ q6asm_read(prtd->audio_client, prtd->stream_id);
break;
default:
@@ -236,7 +237,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
/* rate and channels are sent to audio driver */
if (prtd->state) {
/* clear the previous setup if any */
- q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE);
q6asm_unmap_memory_regions(substream->stream,
prtd->audio_client);
q6routing_stream_close(soc_prtd->dai_link->id,
@@ -255,11 +256,13 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM,
+ ret = q6asm_open_write(prtd->audio_client, prtd->stream_id,
+ FORMAT_LINEAR_PCM,
0, prtd->bits_per_sample);
} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM,
- prtd->bits_per_sample);
+ ret = q6asm_open_read(prtd->audio_client, prtd->stream_id,
+ FORMAT_LINEAR_PCM,
+ prtd->bits_per_sample);
}
if (ret < 0) {
@@ -279,17 +282,19 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
ret = q6asm_media_format_block_multi_ch_pcm(
- prtd->audio_client, runtime->rate,
- runtime->channels, NULL,
+ prtd->audio_client, prtd->stream_id,
+ runtime->rate, runtime->channels, NULL,
prtd->bits_per_sample);
} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
ret = q6asm_enc_cfg_blk_pcm_format_support(prtd->audio_client,
- runtime->rate, runtime->channels,
- prtd->bits_per_sample);
+ prtd->stream_id,
+ runtime->rate,
+ runtime->channels,
+ prtd->bits_per_sample);
/* Queue the buffers */
for (i = 0; i < runtime->periods; i++)
- q6asm_read(prtd->audio_client);
+ q6asm_read(prtd->audio_client, prtd->stream_id);
}
if (ret < 0)
@@ -311,15 +316,18 @@ static int q6asm_dai_trigger(struct snd_soc_component *component,
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id,
+ 0, 0, 0);
break;
case SNDRV_PCM_TRIGGER_STOP:
prtd->state = Q6ASM_STREAM_STOPPED;
- ret = q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+ CMD_EOS);
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- ret = q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+ CMD_PAUSE);
break;
default:
ret = -EINVAL;
@@ -364,6 +372,9 @@ static int q6asm_dai_open(struct snd_soc_component *component,
return ret;
}
+ /* DSP expects stream id from 1 */
+ prtd->stream_id = 1;
+
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
runtime->hw = q6asm_dai_hardware_playback;
else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
@@ -430,7 +441,8 @@ static int q6asm_dai_close(struct snd_soc_component *component,
if (prtd->audio_client) {
if (prtd->state)
- q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_cmd(prtd->audio_client, prtd->stream_id,
+ CMD_CLOSE);
q6asm_unmap_memory_regions(substream->stream,
prtd->audio_client);
@@ -502,8 +514,8 @@ static void compress_event_handler(uint32_t opcode, uint32_t token,
case ASM_CLIENT_EVENT_CMD_RUN_DONE:
spin_lock_irqsave(&prtd->lock, flags);
if (!prtd->bytes_sent) {
- q6asm_write_async(prtd->audio_client, prtd->pcm_count,
- 0, 0, NO_TIMESTAMP);
+ q6asm_write_async(prtd->audio_client, prtd->stream_id,
+ prtd->pcm_count, 0, 0, NO_TIMESTAMP);
prtd->bytes_sent += prtd->pcm_count;
}
@@ -528,7 +540,7 @@ static void compress_event_handler(uint32_t opcode, uint32_t token,
avail = prtd->bytes_received - prtd->bytes_sent;
if (avail >= prtd->pcm_count) {
- q6asm_write_async(prtd->audio_client,
+ q6asm_write_async(prtd->audio_client, prtd->stream_id,
prtd->pcm_count, 0, 0, NO_TIMESTAMP);
prtd->bytes_sent += prtd->pcm_count;
}
@@ -563,6 +575,9 @@ static int q6asm_dai_compr_open(struct snd_soc_component *component,
if (!prtd)
return -ENOMEM;
+ /* DSP expects stream id from 1 */
+ prtd->stream_id = 1;
+
prtd->cstream = stream;
prtd->audio_client = q6asm_audio_client_alloc(dev,
(q6asm_cb)compress_event_handler,
@@ -610,7 +625,8 @@ static int q6asm_dai_compr_free(struct snd_soc_component *component,
if (prtd->audio_client) {
if (prtd->state)
- q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_cmd(prtd->audio_client, prtd->stream_id,
+ CMD_CLOSE);
snd_dma_free_pages(&prtd->dma_buffer);
q6asm_unmap_memory_regions(stream->direction,
@@ -665,8 +681,9 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
prtd->pcm_size = runtime->fragments * runtime->fragment_size;
prtd->bits_per_sample = 16;
if (dir == SND_COMPRESS_PLAYBACK) {
- ret = q6asm_open_write(prtd->audio_client, params->codec.id,
- params->codec.profile, prtd->bits_per_sample);
+ ret = q6asm_open_write(prtd->audio_client, prtd->stream_id,
+ params->codec.id, params->codec.profile,
+ prtd->bits_per_sample);
if (ret < 0) {
dev_err(dev, "q6asm_open_write failed\n");
@@ -700,6 +717,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
flac_cfg.min_frame_size = flac->min_frame_size;
ret = q6asm_stream_media_format_block_flac(prtd->audio_client,
+ prtd->stream_id,
&flac_cfg);
if (ret < 0) {
dev_err(dev, "FLAC CMD Format block failed:%d\n", ret);
@@ -759,10 +777,12 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
if (wma_v9)
ret = q6asm_stream_media_format_block_wma_v9(
- prtd->audio_client, &wma_cfg);
+ prtd->audio_client, prtd->stream_id,
+ &wma_cfg);
else
ret = q6asm_stream_media_format_block_wma_v10(
- prtd->audio_client, &wma_cfg);
+ prtd->audio_client, prtd->stream_id,
+ &wma_cfg);
if (ret < 0) {
dev_err(dev, "WMA9 CMD failed:%d\n", ret);
return -EIO;
@@ -795,6 +815,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
break;
}
ret = q6asm_stream_media_format_block_alac(prtd->audio_client,
+ prtd->stream_id,
&alac_cfg);
if (ret < 0) {
dev_err(dev, "ALAC CMD Format block failed:%d\n", ret);
@@ -819,6 +840,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
ape_cfg.seek_table_present = ape->seek_table_present;
ret = q6asm_stream_media_format_block_ape(prtd->audio_client,
+ prtd->stream_id,
&ape_cfg);
if (ret < 0) {
dev_err(dev, "APE CMD Format block failed:%d\n", ret);
@@ -855,15 +877,18 @@ static int q6asm_dai_compr_trigger(struct snd_soc_component *component,
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id,
+ 0, 0, 0);
break;
case SNDRV_PCM_TRIGGER_STOP:
prtd->state = Q6ASM_STREAM_STOPPED;
- ret = q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+ CMD_EOS);
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- ret = q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+ CMD_PAUSE);
break;
default:
ret = -EINVAL;
@@ -270,7 +270,6 @@ struct audio_client {
wait_queue_head_t cmd_wait;
struct aprv2_ibasic_rsp_result_t result;
int perf_mode;
- int stream_id;
struct q6asm *q6asm;
struct device *dev;
};
@@ -862,8 +861,6 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb,
ac->priv = priv;
ac->io_mode = ASM_SYNC_IO_MODE;
ac->perf_mode = perf_mode;
- /* DSP expects stream id from 1 */
- ac->stream_id = 1;
ac->adev = a->adev;
kref_init(&ac->refcount);
@@ -919,8 +916,9 @@ static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt)
*
* Return: Will be an negative value on error or zero on success
*/
-int q6asm_open_write(struct audio_client *ac, uint32_t format,
- u32 codec_profile, uint16_t bits_per_sample)
+int q6asm_open_write(struct audio_client *ac, uint32_t stream_id,
+ uint32_t format, u32 codec_profile,
+ uint16_t bits_per_sample)
{
struct asm_stream_cmd_open_write_v3 *open;
struct apr_pkt *pkt;
@@ -935,7 +933,7 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
pkt = p;
open = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V3;
open->mode_flags = 0x00;
@@ -998,8 +996,9 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
}
EXPORT_SYMBOL_GPL(q6asm_open_write);
-static int __q6asm_run(struct audio_client *ac, uint32_t flags,
- uint32_t msw_ts, uint32_t lsw_ts, bool wait)
+static int __q6asm_run(struct audio_client *ac, uint32_t stream_id,
+ uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts,
+ bool wait)
{
struct asm_session_cmd_run_v2 *run;
struct apr_pkt *pkt;
@@ -1014,7 +1013,7 @@ static int __q6asm_run(struct audio_client *ac, uint32_t flags,
pkt = p;
run = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_SESSION_CMD_RUN_V2;
run->flags = flags;
@@ -1042,10 +1041,10 @@ static int __q6asm_run(struct audio_client *ac, uint32_t flags,
*
* Return: Will be an negative value on error or zero on success
*/
-int q6asm_run(struct audio_client *ac, uint32_t flags,
+int q6asm_run(struct audio_client *ac, uint32_t stream_id, uint32_t flags,
uint32_t msw_ts, uint32_t lsw_ts)
{
- return __q6asm_run(ac, flags, msw_ts, lsw_ts, true);
+ return __q6asm_run(ac, stream_id, flags, msw_ts, lsw_ts, true);
}
EXPORT_SYMBOL_GPL(q6asm_run);
@@ -1053,16 +1052,17 @@ EXPORT_SYMBOL_GPL(q6asm_run);
* q6asm_run_nowait() - start the audio client withou blocking
*
* @ac: audio client pointer
+ * @stream_id: stream id
* @flags: flags associated with write
* @msw_ts: timestamp msw
* @lsw_ts: timestamp lsw
*
* Return: Will be an negative value on error or zero on success
*/
-int q6asm_run_nowait(struct audio_client *ac, uint32_t flags,
- uint32_t msw_ts, uint32_t lsw_ts)
+int q6asm_run_nowait(struct audio_client *ac, uint32_t stream_id,
+ uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts)
{
- return __q6asm_run(ac, flags, msw_ts, lsw_ts, false);
+ return __q6asm_run(ac, stream_id, flags, msw_ts, lsw_ts, false);
}
EXPORT_SYMBOL_GPL(q6asm_run_nowait);
@@ -1070,6 +1070,7 @@ EXPORT_SYMBOL_GPL(q6asm_run_nowait);
* q6asm_media_format_block_multi_ch_pcm() - setup pcm configuration
*
* @ac: audio client pointer
+ * @stream_id: stream id
* @rate: audio sample rate
* @channels: number of audio channels.
* @channel_map: channel map pointer
@@ -1078,6 +1079,7 @@ EXPORT_SYMBOL_GPL(q6asm_run_nowait);
* Return: Will be an negative value on error or zero on success
*/
int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
+ uint32_t stream_id,
uint32_t rate, uint32_t channels,
u8 channel_map[PCM_MAX_NUM_CHANNEL],
uint16_t bits_per_sample)
@@ -1096,7 +1098,7 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
pkt = p;
fmt = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
@@ -1125,8 +1127,8 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
}
EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm);
-
int q6asm_stream_media_format_block_flac(struct audio_client *ac,
+ uint32_t stream_id,
struct q6asm_flac_cfg *cfg)
{
struct asm_flac_fmt_blk_v2 *fmt;
@@ -1142,7 +1144,7 @@ int q6asm_stream_media_format_block_flac(struct audio_client *ac,
pkt = p;
fmt = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
@@ -1163,6 +1165,7 @@ int q6asm_stream_media_format_block_flac(struct audio_client *ac,
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac);
int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
+ uint32_t stream_id,
struct q6asm_wma_cfg *cfg)
{
struct asm_wmastdv9_fmt_blk_v2 *fmt;
@@ -1178,7 +1181,7 @@ int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
pkt = p;
fmt = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
@@ -1200,6 +1203,7 @@ int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v9);
int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
+ uint32_t stream_id,
struct q6asm_wma_cfg *cfg)
{
struct asm_wmaprov10_fmt_blk_v2 *fmt;
@@ -1215,7 +1219,7 @@ int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
pkt = p;
fmt = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
@@ -1238,6 +1242,7 @@ int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10);
int q6asm_stream_media_format_block_alac(struct audio_client *ac,
+ uint32_t stream_id,
struct q6asm_alac_cfg *cfg)
{
struct asm_alac_fmt_blk_v2 *fmt;
@@ -1253,7 +1258,7 @@ int q6asm_stream_media_format_block_alac(struct audio_client *ac,
pkt = p;
fmt = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
@@ -1279,6 +1284,7 @@ int q6asm_stream_media_format_block_alac(struct audio_client *ac,
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_alac);
int q6asm_stream_media_format_block_ape(struct audio_client *ac,
+ uint32_t stream_id,
struct q6asm_ape_cfg *cfg)
{
struct asm_ape_fmt_blk_v2 *fmt;
@@ -1294,7 +1300,7 @@ int q6asm_stream_media_format_block_ape(struct audio_client *ac,
pkt = p;
fmt = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
@@ -1321,6 +1327,7 @@ EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape);
* q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
*
* @ac: audio client pointer
+ * @stream_id: stream id
* @rate: audio sample rate
* @channels: number of audio channels.
* @bits_per_sample: bits per sample
@@ -1328,7 +1335,9 @@ EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape);
* Return: Will be an negative value on error or zero on success
*/
int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
- uint32_t rate, uint32_t channels, uint16_t bits_per_sample)
+ uint32_t stream_id, uint32_t rate,
+ uint32_t channels,
+ uint16_t bits_per_sample)
{
struct asm_multi_channel_pcm_enc_cfg_v2 *enc_cfg;
struct apr_pkt *pkt;
@@ -1344,7 +1353,7 @@ int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
pkt = p;
enc_cfg = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
enc_cfg->encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2;
@@ -1376,10 +1385,11 @@ EXPORT_SYMBOL_GPL(q6asm_enc_cfg_blk_pcm_format_support);
* q6asm_read() - read data of period size from audio client
*
* @ac: audio client pointer
+ * @stream_id: stream id
*
* Return: Will be an negative value on error or zero on success
*/
-int q6asm_read(struct audio_client *ac)
+int q6asm_read(struct audio_client *ac, uint32_t stream_id)
{
struct asm_data_cmd_read_v2 *read;
struct audio_port_data *port;
@@ -1400,7 +1410,7 @@ int q6asm_read(struct audio_client *ac)
spin_lock_irqsave(&ac->lock, flags);
port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, stream_id);
ab = &port->buf[port->dsp_buf];
pkt->hdr.opcode = ASM_DATA_CMD_READ_V2;
read->buf_addr_lsw = lower_32_bits(ab->phys);
@@ -1428,7 +1438,7 @@ int q6asm_read(struct audio_client *ac)
}
EXPORT_SYMBOL_GPL(q6asm_read);
-static int __q6asm_open_read(struct audio_client *ac,
+static int __q6asm_open_read(struct audio_client *ac, uint32_t stream_id,
uint32_t format, uint16_t bits_per_sample)
{
struct asm_stream_cmd_open_read_v3 *open;
@@ -1444,7 +1454,7 @@ static int __q6asm_open_read(struct audio_client *ac,
pkt = p;
open = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_READ_V3;
/* Stream prio : High, provide meta info with encoded frames */
open->src_endpointype = ASM_END_POINT_DEVICE_MATRIX;
@@ -1475,15 +1485,16 @@ static int __q6asm_open_read(struct audio_client *ac,
* q6asm_open_read() - Open audio client for reading
*
* @ac: audio client pointer
+ * @stream_id: stream id
* @format: audio sample format
* @bits_per_sample: bits per sample
*
* Return: Will be an negative value on error or zero on success
*/
-int q6asm_open_read(struct audio_client *ac, uint32_t format,
- uint16_t bits_per_sample)
+int q6asm_open_read(struct audio_client *ac, uint32_t stream_id,
+ uint32_t format, uint16_t bits_per_sample)
{
- return __q6asm_open_read(ac, format, bits_per_sample);
+ return __q6asm_open_read(ac, stream_id, format, bits_per_sample);
}
EXPORT_SYMBOL_GPL(q6asm_open_read);
@@ -1491,6 +1502,7 @@ EXPORT_SYMBOL_GPL(q6asm_open_read);
* q6asm_write_async() - non blocking write
*
* @ac: audio client pointer
+ * @stream_id: stream id
* @len: length in bytes
* @msw_ts: timestamp msw
* @lsw_ts: timestamp lsw
@@ -1498,8 +1510,8 @@ EXPORT_SYMBOL_GPL(q6asm_open_read);
*
* Return: Will be an negative value on error or zero on success
*/
-int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
- uint32_t lsw_ts, uint32_t wflags)
+int q6asm_write_async(struct audio_client *ac, uint32_t stream_id, uint32_t len,
+ uint32_t msw_ts, uint32_t lsw_ts, uint32_t wflags)
{
struct asm_data_cmd_write_v2 *write;
struct audio_port_data *port;
@@ -1520,7 +1532,7 @@ int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
spin_lock_irqsave(&ac->lock, flags);
port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, stream_id);
ab = &port->buf[port->dsp_buf];
pkt->hdr.token = port->dsp_buf;
@@ -1567,9 +1579,9 @@ static void q6asm_reset_buf_state(struct audio_client *ac)
spin_unlock_irqrestore(&ac->lock, flags);
}
-static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait)
+static int __q6asm_cmd(struct audio_client *ac, uint32_t stream_id, int cmd,
+ bool wait)
{
- int stream_id = ac->stream_id;
struct apr_pkt pkt;
int rc;
@@ -1616,13 +1628,14 @@ static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait)
* q6asm_cmd() - run cmd on audio client
*
* @ac: audio client pointer
+ * @stream_id: stream id
* @cmd: command to run on audio client.
*
* Return: Will be an negative value on error or zero on success
*/
-int q6asm_cmd(struct audio_client *ac, int cmd)
+int q6asm_cmd(struct audio_client *ac, uint32_t stream_id, int cmd)
{
- return __q6asm_cmd(ac, cmd, true);
+ return __q6asm_cmd(ac, stream_id, cmd, true);
}
EXPORT_SYMBOL_GPL(q6asm_cmd);
@@ -1630,13 +1643,14 @@ EXPORT_SYMBOL_GPL(q6asm_cmd);
* q6asm_cmd_nowait() - non blocking, run cmd on audio client
*
* @ac: audio client pointer
+ * @stream_id: stream id
* @cmd: command to run on audio client.
*
* Return: Will be an negative value on error or zero on success
*/
-int q6asm_cmd_nowait(struct audio_client *ac, int cmd)
+int q6asm_cmd_nowait(struct audio_client *ac, uint32_t stream_id, int cmd)
{
- return __q6asm_cmd(ac, cmd, false);
+ return __q6asm_cmd(ac, stream_id, cmd, false);
}
EXPORT_SYMBOL_GPL(q6asm_cmd_nowait);
@@ -93,37 +93,47 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev,
q6asm_cb cb, void *priv,
int session_id, int perf_mode);
void q6asm_audio_client_free(struct audio_client *ac);
-int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
- uint32_t lsw_ts, uint32_t flags);
-int q6asm_open_write(struct audio_client *ac, uint32_t format,
- u32 codec_profile, uint16_t bits_per_sample);
-
-int q6asm_open_read(struct audio_client *ac, uint32_t format,
+int q6asm_write_async(struct audio_client *ac, uint32_t stream_id, uint32_t len,
+ uint32_t msw_ts, uint32_t lsw_ts, uint32_t flags);
+int q6asm_open_write(struct audio_client *ac, uint32_t stream_id,
+ uint32_t format, u32 codec_profile,
uint16_t bits_per_sample);
+
+int q6asm_open_read(struct audio_client *ac, uint32_t stream_id,
+ uint32_t format, uint16_t bits_per_sample);
int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
- uint32_t rate, uint32_t channels, uint16_t bits_per_sample);
-int q6asm_read(struct audio_client *ac);
+ uint32_t stream_id, uint32_t rate,
+ uint32_t channels,
+ uint16_t bits_per_sample);
+
+int q6asm_read(struct audio_client *ac, uint32_t stream_id);
int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
+ uint32_t stream_id,
uint32_t rate, uint32_t channels,
u8 channel_map[PCM_MAX_NUM_CHANNEL],
uint16_t bits_per_sample);
int q6asm_stream_media_format_block_flac(struct audio_client *ac,
+ uint32_t stream_id,
struct q6asm_flac_cfg *cfg);
int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
+ uint32_t stream_id,
struct q6asm_wma_cfg *cfg);
int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
+ uint32_t stream_id,
struct q6asm_wma_cfg *cfg);
int q6asm_stream_media_format_block_alac(struct audio_client *ac,
+ uint32_t stream_id,
struct q6asm_alac_cfg *cfg);
int q6asm_stream_media_format_block_ape(struct audio_client *ac,
+ uint32_t stream_id,
struct q6asm_ape_cfg *cfg);
-int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
- uint32_t lsw_ts);
-int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
- uint32_t lsw_ts);
-int q6asm_cmd(struct audio_client *ac, int cmd);
-int q6asm_cmd_nowait(struct audio_client *ac, int cmd);
+int q6asm_run(struct audio_client *ac, uint32_t stream_id, uint32_t flags,
+ uint32_t msw_ts, uint32_t lsw_ts);
+int q6asm_run_nowait(struct audio_client *ac, uint32_t stream_id,
+ uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts);
+int q6asm_cmd(struct audio_client *ac, uint32_t stream_id, int cmd);
+int q6asm_cmd_nowait(struct audio_client *ac, uint32_t stream_id, int cmd);
int q6asm_get_session_id(struct audio_client *ac);
int q6asm_map_memory_regions(unsigned int dir,
struct audio_client *ac,
Each ASM session can have multiple streams attached to it, current design was to allow only one static stream id 1 per each session. However for use-case like gapless, we would need 2 streams to open per session. This patch converts all the q6asm apis to take stream id as argument to allow multiple streams to open on a single session, This is useful for gapless playback cases. Now the dai driver can specify which stream id for each command. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> --- sound/soc/qcom/qdsp6/q6asm-dai.c | 79 +++++++++++++++++---------- sound/soc/qcom/qdsp6/q6asm.c | 92 ++++++++++++++++++-------------- sound/soc/qcom/qdsp6/q6asm.h | 38 ++++++++----- 3 files changed, 129 insertions(+), 80 deletions(-)